LIVE555中RTSP客户端接收媒体流分析及测试代码

水深无声 2022-01-21 09:57 374阅读 0赞

LIVE555中testProgs目录下的testRTSPClient.cpp代码用于测试接收RTSP URL指定的媒体流，向服务器端发送的命令包括：DESCRIBE、SETUP、PLAY、TERADOWN。

1. 设置使用环境：new一个BasicTaskScheduler对象；new一个BasicUsageEnvironment对象；

2. new一个RTSPClient对象；

3. 向服务器发送一个RTSP “DESCRIBE”命令，以获取stream的SDP描述(description)，异步操作: RTSPClient::sendDescribeCommand；

4. 处理从服务器端返回的“DESCRIBE”响应：SDP描述，通过SDP描述创建一个会话MediaSession，接着为此会话创建数据源对象MediaSubsessionIterator，continueAfterDESCRIBE为回调函数；

5. 向服务器发送一个RTSP “SETUP”命令，异步操作，在setupNextSubsession函数中：RTSPClient::sendSetupCommand;

6. 处理从服务器端返回的”SETUP”响应：创建一个MediaSink对象DummySink对象用于接收流数据，continueAfterSETUP为回调函数；

7. 向服务器发送一个RTSP “PLAY”命令，异步操作，在setupNextSubsession函数中：RTSPClient::sendPlayCommand；

8. 处理从服务器端返回的”PLAY”响应：设置一个定时器用于结束接收流，continueAfterPLAY为回调函数；

9. 接收流数据，包括音频流和视频流，可通过MediaSubsession::mediumName()函数来判断此流是视频流(video)还是音频流(audio)；可通过MediaSubsession::codecName()函数来判断是采用的哪种编码标准；每次接收流的大小是不同的，可通过frameSize来获取当次接收流的字节数，流数据在fReceiveBuffer中获取；在每次接收视频流数据的开头需要加入0x00000001，这样FFmpeg才能成功解码；

10. 当出现异常情况或正常结束接收流时会向服务器发送一个RTSP “TERADOWN”命令，异步操作，在shutdownStream中：RTSPClient::sendTeardownCommand。

注：

1. 每一路都需要单独创建一个RTSPClient对象。

2. 可以在某个时间点上设置eventLoopWatchVariable为非0值来正常结束程序。

3. MediaSource是所有Source的基类；MediaSink是所有Sink的基类；它们均又继承自Media类。MediaSession用于表示一个RTP会话，一个MediaSession可能包含多个子会话(MediaSubSession)。Source和Sink通过RTP子会话联系在一起。Source：发送端，流的起点。Sink：接收端，流的终点，数据流保存到文件或显示，数据通过Source传递到Sink。每一个MediaSubSession内部维护一个Source和Sink。

以下是基于testRTSPClient.cpp的测试代码：

#include "funset.hpp"
    #include <iostream>
    #include <thread>
    #include <chrono>
    #include <liveMedia.hh>
    #include <BasicUsageEnvironment.hh>
    
    namespace {
    
    // Forward function definitions:
    // RTSP 'response handlers':
    void continueAfterDESCRIBE(RTSPClient* rtspClient, int resultCode, char* resultString);
    void continueAfterSETUP(RTSPClient* rtspClient, int resultCode, char* resultString);
    void continueAfterPLAY(RTSPClient* rtspClient, int resultCode, char* resultString);
    
    // Other event handler functions:
    // called when a stream's subsession (e.g., audio or video substream) ends
    void subsessionAfterPlaying(void* clientData);
    // called when a RTCP "BYE" is received for a subsession
    void subsessionByeHandler(void* clientData, char const* reason);
    // called at the end of a stream's expected duration (if the stream has not already signaled its end using a RTCP "BYE")
    void streamTimerHandler(void* clientData);
    
    // The main streaming routine (for each "rtsp://" URL):
    void openURL(UsageEnvironment& env, char const* progName, char const* rtspURL);
    
    // Used to iterate through each stream's 'subsessions', setting up each one:
    void setupNextSubsession(RTSPClient* rtspClient);
    
    // Used to shut down and close a stream (including its "RTSPClient" object):
    void shutdownStream(RTSPClient* rtspClient, int exitCode = 1);
    
    // A function that outputs a string that identifies each stream (for debugging output).  Modify this if you wish:
    UsageEnvironment& operator<<(UsageEnvironment& env, const RTSPClient& rtspClient)
    {
    	return env << "[URL:\"" << rtspClient.url() << "\"]: ";
    }
    
    // A function that outputs a string that identifies each subsession (for debugging output).  Modify this if you wish:
    UsageEnvironment& operator<<(UsageEnvironment& env, const MediaSubsession& subsession)
    {
    	return env << subsession.mediumName() << "/" << subsession.codecName();
    }
    
    char eventLoopWatchVariable = 0;
    
    // Define a class to hold per-stream state that we maintain throughout each stream's lifetime:
    class StreamClientState {
    public:
    	StreamClientState();
    	virtual ~StreamClientState();
    
    public:
    	MediaSubsessionIterator* iter;
    	MediaSession* session;
    	MediaSubsession* subsession;
    	TaskToken streamTimerTask;
    	double duration;
    };
    
    // If you're streaming just a single stream (i.e., just from a single URL, once), then you can define and use just a single
    // "StreamClientState" structure, as a global variable in your application.  However, because - in this demo application - we're
    // showing how to play multiple streams, concurrently, we can't do that.  Instead, we have to have a separate "StreamClientState"
    // structure for each "RTSPClient".  To do this, we subclass "RTSPClient", and add a "StreamClientState" field to the subclass:
    class ourRTSPClient : public RTSPClient {
    public:
    	static ourRTSPClient* createNew(UsageEnvironment& env, char const* rtspURL,
    		int verbosityLevel = 0, char const* applicationName = NULL, portNumBits tunnelOverHTTPPortNum = 0);
    
    protected:
    	ourRTSPClient(UsageEnvironment& env, char const* rtspURL, int verbosityLevel, char const* applicationName, portNumBits tunnelOverHTTPPortNum);
    	// called only by createNew();
    	virtual ~ourRTSPClient();
    
    public:
    	StreamClientState scs;
    };
    
    // Define a data sink (a subclass of "MediaSink") to receive the data for each subsession (i.e., each audio or video 'substream').
    // In practice, this might be a class (or a chain of classes) that decodes and then renders the incoming audio or video.
    // Or it might be a "FileSink", for outputting the received data into a file (as is done by the "openRTSP" application).
    // In this example code, however, we define a simple 'dummy' sink that receives incoming data, but does nothing with it.
    class DummySink : public MediaSink {
    public:
    	static DummySink* createNew(UsageEnvironment& env,
    		MediaSubsession& subsession, // identifies the kind of data that's being received
    		char const* streamId = NULL); // identifies the stream itself (optional)
    
    private:
    	DummySink(UsageEnvironment& env, MediaSubsession& subsession, char const* streamId);
    	// called only by "createNew()"
    	virtual ~DummySink();
    
    	static void afterGettingFrame(void* clientData, unsigned frameSize, unsigned numTruncatedBytes,
    					struct timeval presentationTime, unsigned durationInMicroseconds);
    	void afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes, struct timeval presentationTime, unsigned durationInMicroseconds);
    
    private:
    	// redefined virtual functions:
    	virtual Boolean continuePlaying();
    
    private:
    	u_int8_t* fReceiveBuffer;
    	MediaSubsession& fSubsession;
    	char* fStreamId;
    };
    
    #define RTSP_CLIENT_VERBOSITY_LEVEL 1 // by default, print verbose output from each "RTSPClient"
    
    static unsigned rtspClientCount = 0; // Counts how many streams (i.e., "RTSPClient"s) are currently in use.
    
    void openURL(UsageEnvironment& env, char const* progName, char const* rtspURL)
    {
    	// Begin by creating a "RTSPClient" object.  Note that there is a separate "RTSPClient" object for each stream that we wish
    	// to receive (even if more than stream uses the same "rtsp://" URL).
    	RTSPClient* rtspClient = ourRTSPClient::createNew(env, rtspURL, RTSP_CLIENT_VERBOSITY_LEVEL, progName);
    	if (rtspClient == NULL) {
    		env << "Failed to create a RTSP client for URL \"" << rtspURL << "\": " << env.getResultMsg() << "\n";
    		return;
    	}
    
    	++rtspClientCount;
    
    	// Next, send a RTSP "DESCRIBE" command, to get a SDP description for the stream.
    	// Note that this command - like all RTSP commands - is sent asynchronously; we do not block, waiting for a response.
    	// Instead, the following function call returns immediately, and we handle the RTSP response later, from within the event loop:
    	rtspClient->sendDescribeCommand(continueAfterDESCRIBE);
    }
    
    // Implementation of the RTSP 'response handlers':
    void continueAfterDESCRIBE(RTSPClient* rtspClient, int resultCode, char* resultString)
    {
    	do {
    		UsageEnvironment& env = rtspClient->envir(); // alias
    		StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
    
    		if (resultCode != 0) {
    			env << *rtspClient << "Failed to get a SDP description: " << resultString << ", resultCode: " << resultCode << "\n";
    			delete[] resultString;
    			break;
    		}
    
    		char* const sdpDescription = resultString;
    		env << *rtspClient << "Got a SDP description:\n" << sdpDescription << "\n";
    
    		// Create a media session object from this SDP description:
    		scs.session = MediaSession::createNew(env, sdpDescription);
    		delete[] sdpDescription; // because we don't need it anymore
    		if (scs.session == NULL) {
    			env << *rtspClient << "Failed to create a MediaSession object from the SDP description: " << env.getResultMsg() << "\n";
    			break;
    		} else if (!scs.session->hasSubsessions()) {
    			env << *rtspClient << "This session has no media subsessions (i.e., no \"m=\" lines)\n";
    			break;
    		}
    
    		// Then, create and set up our data source objects for the session.  We do this by iterating over the session's 'subsessions',
    		// calling "MediaSubsession::initiate()", and then sending a RTSP "SETUP" command, on each one.
    		// (Each 'subsession' will have its own data source.)
    		scs.iter = new MediaSubsessionIterator(*scs.session);
    		setupNextSubsession(rtspClient);
    		return;
    	} while (0);
    
    	// An unrecoverable error occurred with this stream.
    	shutdownStream(rtspClient);
    }
    
    // By default, we request that the server stream its data using RTP/UDP.
    // If, instead, you want to request that the server stream via RTP-over-TCP, change the following to True:
    #define REQUEST_STREAMING_OVER_TCP False
    
    void setupNextSubsession(RTSPClient* rtspClient)
    {
    	UsageEnvironment& env = rtspClient->envir(); // alias
    	StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
    
    	scs.subsession = scs.iter->next();
    	if (scs.subsession != NULL) {
    		if (!scs.subsession->initiate()) {
    			env << *rtspClient << "Failed to initiate the \"" << *scs.subsession << "\" subsession: " << env.getResultMsg() << "\n";
    			setupNextSubsession(rtspClient); // give up on this subsession; go to the next one
    		} else {
    			env << *rtspClient << "Initiated the \"" << *scs.subsession << "\" subsession (";
    			if (scs.subsession->rtcpIsMuxed()) {
    				env << "client port " << scs.subsession->clientPortNum();
    			} else {
    				env << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum() + 1;
    			}
    			env << ")\n";
    
    			// Continue setting up this subsession, by sending a RTSP "SETUP" command:
    			rtspClient->sendSetupCommand(*scs.subsession, continueAfterSETUP, False, REQUEST_STREAMING_OVER_TCP);
    		}
    		return;
    	}
    
    	// We've finished setting up all of the subsessions.  Now, send a RTSP "PLAY" command to start the streaming:
    	if (scs.session->absStartTime() != NULL) {
    		// Special case: The stream is indexed by 'absolute' time, so send an appropriate "PLAY" command:
    		rtspClient->sendPlayCommand(*scs.session, continueAfterPLAY, scs.session->absStartTime(), scs.session->absEndTime());
    	} else {
    		scs.duration = scs.session->playEndTime() - scs.session->playStartTime();
    		rtspClient->sendPlayCommand(*scs.session, continueAfterPLAY);
    	}
    }
    
    void continueAfterSETUP(RTSPClient* rtspClient, int resultCode, char* resultString)
    {
    	do {
    		UsageEnvironment& env = rtspClient->envir(); // alias
    		StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
    
    		if (resultCode != 0) {
    			env << *rtspClient << "Failed to set up the \"" << *scs.subsession << "\" subsession: " << resultString << ", resultCode: " << resultCode << "\n";
    			break;
    		}
    
    		env << *rtspClient << "Set up the \"" << *scs.subsession << "\" subsession (";
    		if (scs.subsession->rtcpIsMuxed()) {
    			env << "client port " << scs.subsession->clientPortNum();
    		} else {
    			env << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum() + 1;
    		}
    		env << ")\n";
    
    		// Having successfully setup the subsession, create a data sink for it, and call "startPlaying()" on it.
    		// (This will prepare the data sink to receive data; the actual flow of data from the client won't start happening until later,
    		// after we've sent a RTSP "PLAY" command.)
    		scs.subsession->sink = DummySink::createNew(env, *scs.subsession, rtspClient->url());
    		// perhaps use your own custom "MediaSink" subclass instead
    		if (scs.subsession->sink == NULL) {
    			env << *rtspClient << "Failed to create a data sink for the \"" << *scs.subsession
    				<< "\" subsession: " << env.getResultMsg() << "\n";
    			break;
    		}
    
    		env << *rtspClient << "Created a data sink for the \"" << *scs.subsession << "\" subsession\n";
    		scs.subsession->miscPtr = rtspClient; // a hack to let subsession handler functions get the "RTSPClient" from the subsession 
    		scs.subsession->sink->startPlaying(*(scs.subsession->readSource()), subsessionAfterPlaying, scs.subsession);
    		// Also set a handler to be called if a RTCP "BYE" arrives for this subsession:
    		if (scs.subsession->rtcpInstance() != NULL) {
    			scs.subsession->rtcpInstance()->setByeWithReasonHandler(subsessionByeHandler, scs.subsession);
    		}
    	} while (0);
    	delete[] resultString;
    
    	// Set up the next subsession, if any:
    	setupNextSubsession(rtspClient);
    }
    
    void continueAfterPLAY(RTSPClient* rtspClient, int resultCode, char* resultString)
    {
    	Boolean success = False;
    
    	do {
    		UsageEnvironment& env = rtspClient->envir(); // alias
    		StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
    
    		if (resultCode != 0) {
    			env << *rtspClient << "Failed to start playing session: " << resultString << ", resultCode: " << resultCode << "\n";
    			break;
    		}
    
    		// Set a timer to be handled at the end of the stream's expected duration (if the stream does not already signal its end
    		// using a RTCP "BYE").  This is optional.  If, instead, you want to keep the stream active - e.g., so you can later
    		// 'seek' back within it and do another RTSP "PLAY" - then you can omit this code.
    		// (Alternatively, if you don't want to receive the entire stream, you could set this timer for some shorter value.)
    		if (scs.duration > 0) {
    			unsigned const delaySlop = 2; // number of seconds extra to delay, after the stream's expected duration.  (This is optional.)
    			scs.duration += delaySlop;
    			unsigned uSecsToDelay = (unsigned)(scs.duration * 1000000);
    			scs.streamTimerTask = env.taskScheduler().scheduleDelayedTask(uSecsToDelay, (TaskFunc*)streamTimerHandler, rtspClient);
    		}
    
    		env << *rtspClient << "Started playing session";
    		if (scs.duration > 0) {
    			env << " (for up to " << scs.duration << " seconds)";
    		}
    		env << "...\n";
    
    		success = True;
    	} while (0);
    	delete[] resultString;
    
    	if (!success) {
    		// An unrecoverable error occurred with this stream.
    		shutdownStream(rtspClient);
    	}
    }
    
    // Implementation of the other event handlers:
    void subsessionAfterPlaying(void* clientData)
    {
    	MediaSubsession* subsession = (MediaSubsession*)clientData;
    	RTSPClient* rtspClient = (RTSPClient*)(subsession->miscPtr);
    
    	// Begin by closing this subsession's stream:
    	Medium::close(subsession->sink);
    	subsession->sink = NULL;
    
    	// Next, check whether *all* subsessions' streams have now been closed:
    	MediaSession& session = subsession->parentSession();
    	MediaSubsessionIterator iter(session);
    	while ((subsession = iter.next()) != NULL) {
    		if (subsession->sink != NULL) return; // this subsession is still active
    	}
    
    	// All subsessions' streams have now been closed, so shutdown the client:
    	shutdownStream(rtspClient);
    }
    
    void subsessionByeHandler(void* clientData, char const* reason)
    {
    	MediaSubsession* subsession = (MediaSubsession*)clientData;
    	RTSPClient* rtspClient = (RTSPClient*)subsession->miscPtr;
    	UsageEnvironment& env = rtspClient->envir(); // alias
    
    	env << *rtspClient << "Received RTCP \"BYE\"";
    	if (reason != NULL) {
    		env << " (reason:\"" << reason << "\")";
    		delete[] reason;
    	}
    	env << " on \"" << *subsession << "\" subsession\n";
    
    	// Now act as if the subsession had closed:
    	subsessionAfterPlaying(subsession);
    }
    
    void streamTimerHandler(void* clientData)
    {
    	ourRTSPClient* rtspClient = (ourRTSPClient*)clientData;
    	StreamClientState& scs = rtspClient->scs; // alias
    
    	scs.streamTimerTask = NULL;
    
    	// Shut down the stream:
    	shutdownStream(rtspClient);
    }
    
    void shutdownStream(RTSPClient* rtspClient, int exitCode)
    {
    	UsageEnvironment& env = rtspClient->envir(); // alias
    	StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
    
    	// First, check whether any subsessions have still to be closed:
    	if (scs.session != NULL) {
    		Boolean someSubsessionsWereActive = False;
    		MediaSubsessionIterator iter(*scs.session);
    		MediaSubsession* subsession;
    
    		while ((subsession = iter.next()) != NULL) {
    			if (subsession->sink != NULL) {
    				Medium::close(subsession->sink);
    				subsession->sink = NULL;
    
    				if (subsession->rtcpInstance() != NULL) {
    					subsession->rtcpInstance()->setByeHandler(NULL, NULL); // in case the server sends a RTCP "BYE" while handling "TEARDOWN"
    				}
    
    				someSubsessionsWereActive = True;
    			}
    		}
    
    		if (someSubsessionsWereActive) {
    			// Send a RTSP "TEARDOWN" command, to tell the server to shutdown the stream.
    			// Don't bother handling the response to the "TEARDOWN".
    			rtspClient->sendTeardownCommand(*scs.session, NULL);
    		}
    	}
    
    	env << *rtspClient << "Closing the stream.\n";
    	Medium::close(rtspClient);
    	// Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.
    
    	if (--rtspClientCount == 0) {
    		// The final stream has ended, so exit the application now.
    		// (Of course, if you're embedding this code into your own application, you might want to comment this out,
    		// and replace it with "eventLoopWatchVariable = 1;", so that we leave the LIVE555 event loop, and continue running "main()".)
    		//exit(exitCode);
    		eventLoopWatchVariable = 1;
    	}
    }
    
    // Implementation of "ourRTSPClient":
    ourRTSPClient* ourRTSPClient::createNew(UsageEnvironment& env, char const* rtspURL,
    	int verbosityLevel, char const* applicationName, portNumBits tunnelOverHTTPPortNum)
    {
    	return new ourRTSPClient(env, rtspURL, verbosityLevel, applicationName, tunnelOverHTTPPortNum);
    }
    
    ourRTSPClient::ourRTSPClient(UsageEnvironment& env, char const* rtspURL,
    	int verbosityLevel, char const* applicationName, portNumBits tunnelOverHTTPPortNum)
    	: RTSPClient(env, rtspURL, verbosityLevel, applicationName, tunnelOverHTTPPortNum, -1)
    {
    }
    
    ourRTSPClient::~ourRTSPClient()
    {
    }
    
    // Implementation of "StreamClientState":
    StreamClientState::StreamClientState()
    	: iter(NULL), session(NULL), subsession(NULL), streamTimerTask(NULL), duration(0.0)
    {
    }
    
    StreamClientState::~StreamClientState()
    {
    	delete iter;
    	if (session != NULL) {
    		// We also need to delete "session", and unschedule "streamTimerTask" (if set)
    		UsageEnvironment& env = session->envir(); // alias
    
    		env.taskScheduler().unscheduleDelayedTask(streamTimerTask);
    		Medium::close(session);
    	}
    }
    
    // Implementation of "DummySink":
    // Even though we're not going to be doing anything with the incoming data, we still need to receive it.
    // Define the size of the buffer that we'll use:
    #define DUMMY_SINK_RECEIVE_BUFFER_SIZE 100000
    
    DummySink* DummySink::createNew(UsageEnvironment& env, MediaSubsession& subsession, char const* streamId)
    {
    	return new DummySink(env, subsession, streamId);
    }
    
    DummySink::DummySink(UsageEnvironment& env, MediaSubsession& subsession, char const* streamId)
    	: MediaSink(env), fSubsession(subsession)
    {
    	fStreamId = strDup(streamId);
    	fReceiveBuffer = new u_int8_t[DUMMY_SINK_RECEIVE_BUFFER_SIZE];
    }
    
    DummySink::~DummySink()
    {
    	delete[] fReceiveBuffer;
    	delete[] fStreamId;
    }
    
    void DummySink::afterGettingFrame(void* clientData, unsigned frameSize, unsigned numTruncatedBytes, struct timeval presentationTime, unsigned durationInMicroseconds)
    {
    	DummySink* sink = (DummySink*)clientData;
    	sink->afterGettingFrame(frameSize, numTruncatedBytes, presentationTime, durationInMicroseconds);
    }
    
    // If you don't want to see debugging output for each received frame, then comment out the following line:
    #define DEBUG_PRINT_EACH_RECEIVED_FRAME 1
    
    void DummySink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes, struct timeval presentationTime, unsigned /*durationInMicroseconds*/)
    {
    	// We've just received a frame of data.  (Optionally) print out information about it:
    #ifdef DEBUG_PRINT_EACH_RECEIVED_FRAME
    	if (fStreamId != NULL) envir() << "Stream \"" << fStreamId << "\"; ";
    	envir() << fSubsession.mediumName() << "/" << fSubsession.codecName() << ":\tReceived " << frameSize << " bytes";
    	if (numTruncatedBytes > 0) envir() << " (with " << numTruncatedBytes << " bytes truncated)";
    	char uSecsStr[6 + 1]; // used to output the 'microseconds' part of the presentation time
    	sprintf(uSecsStr, "%06u", (unsigned)presentationTime.tv_usec);
    	envir() << ".\tPresentation time: " << (int)presentationTime.tv_sec << "." << uSecsStr;
    	if (fSubsession.rtpSource() != NULL && !fSubsession.rtpSource()->hasBeenSynchronizedUsingRTCP()) {
    		envir() << "!"; // mark the debugging output to indicate that this presentation time is not RTCP-synchronized
    	}
    #ifdef DEBUG_PRINT_NPT
    	envir() << "\tNPT: " << fSubsession.getNormalPlayTime(presentationTime);
    #endif
    	envir() << "\n";
    #endif
    
    	// Then continue, to request the next frame of data:
    	continuePlaying();
    }
    
    Boolean DummySink::continuePlaying()
    {
    	if (fSource == NULL) return False; // sanity check (should not happen)
    
    	// Request the next frame of data from our input source.  "afterGettingFrame()" will get called later, when it arrives:
    	fSource->getNextFrame(fReceiveBuffer, DUMMY_SINK_RECEIVE_BUFFER_SIZE, afterGettingFrame, this, onSourceClosure, this);
    	return True;
    }
    
    void shutdown_rtsp_client_stream()
    {
    	while (1) {
    		std::this_thread::sleep_for(std::chrono::seconds(2));
    		eventLoopWatchVariable = 1;
    		break;
    	}
    }
    
    } // namespace
    
    int test_live555_rtsp_client()
    {
    	// reference: live/testProgs/testRTSPClient.cpp
    
    	// Begin by setting up our usage environment:
    	TaskScheduler* scheduler = BasicTaskScheduler::createNew();
    	UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
    
    	// We need at least one "rtsp://" URL argument:
    	int argc = 2;
    	char* argv[2] = { "test_live555_rtsp_client", "rtsp://184.72.239.149/vod/mp4://BigBuckBunny_175k.mov"};
    
    	// There are argc-1 URLs: argv[1] through argv[argc-1].  Open and start streaming each one:
    	for (int i = 1; i <= argc - 1; ++i) {
    		openURL(*env, argv[0], argv[i]);
    	}
    
    	std::thread th(shutdown_rtsp_client_stream);
    
    	// All subsequent activity takes place within the event loop:
    	// This function call does not return, unless, at some point in time, "eventLoopWatchVariable" gets set to something non-zero.
    	env->taskScheduler().doEventLoop(&eventLoopWatchVariable);
    
    	th.join();
    
    	// If you choose to continue the application past this point (i.e., if you comment out the "return 0;" statement above),
    	// and if you don't intend to do anything more with the "TaskScheduler" and "UsageEnvironment" objects,
    	// then you can also reclaim the (small) memory used by these objects by uncommenting the following code:
    	env->reclaim(); env = NULL;
    	delete scheduler; scheduler = NULL;
    
    	return 0;
    }

执行结果如下：

![watermark_type_ZmFuZ3poZW5naGVpdGk_shadow_10_text_aHR0cHM6Ly9ibG9nLmNzZG4ubmV0L2ZlbmdiaW5nY2h1bg_size_16_color_FFFFFF_t_70][]

**GitHub**：[https://github.com/fengbingchun/OpenCV\_Test][https_github.com_fengbingchun_OpenCV_Test]

[watermark_type_ZmFuZ3poZW5naGVpdGk_shadow_10_text_aHR0cHM6Ly9ibG9nLmNzZG4ubmV0L2ZlbmdiaW5nY2h1bg_size_16_color_FFFFFF_t_70]: /images/20220121/5e70aff270ed44f892c7fcc61b4b9657.png
[https_github.com_fengbingchun_OpenCV_Test]: https://github.com/fengbingchun/OpenCV_Test